This article describes the setup, operation, and operation of OpenStage SIP and OpenScape Desk Phone IP phones in an Asterisk telephony environment. For a detailed HowTo, please see HowTo_OpenStage_Asterisk.

Common telephony features are supported out of the box, such as call transfer, call forwarding, consultation, voicemail, and many more. To add more functionality in order to build a feature-rich communication system, two additional interfaces can be used by developers:

The OpenStage_WPI (WorkPoint Interface) provides an open, XML-based provisioning interface to support mass deployment and to enable automated updates and configuration.

OpenStage 60 and 80 and OpenScape Desk Phone IP 55G provide an XML-based application interface which allows for developing graphical applications hosted on a remote server. Besides displaying information, sending data, and controlling all sorts of remote processes, these applications have the capability of controlling calls.

External Power Supply

For an OpenStage 60/80 G or an OpenScape Desk Phone IP 55G with a 2nd Key Module, an external power unit is required.
The order no. for the plug-in power supply is region specific:

EU: C39280-Z4-C510

UK: C39280-Z4-C512

USA: C39280-Z4-C511

Energy Saving

OpenStage phones and OpenScape Desk Phones offer an energy saving mode. The display backlight is switched off after a configurable timeout.
With OpenStage 40, the main display and key module backlight will be switched off after 90 seconds of inactivity (firmware version V2R0 onwards). Readability even without backlight is ensured by the transflective display.
With OpenStage 60 and 80 or OpenScape Desk Phone IP 55G, the timer is configurable by the administrator (Local Functions > Energy saving); the timeout ranges between 2 and 8 hours.

Connecting to an IP Network

802.1x

OpenStage phones and OpenScape Desk Phones support 802.1x EAP-TLS. Certificates for authentication can be downloaded via the WPI.

LLDP-MED

OpenStage SIP phones and OpenScape Desk Phones IP support the layer 2 protocol LLDP-MED (Link Layer Discovery Protocol-Media Endpoint Discovery). When an OpenStage phone or an OpenScape Desk Phone is connected to a switch with LLDP-MED capabilities, the phone is able to

advertise and receive a VLAN ID,

advertise and receive QoS parameters,

advertise the power requirements to the LAN access switch by means of an "Extended Power via MDI" TLV.

LLDEP-MED usage is configured in the administratio menu under Network > IP configuration.

DHCP

The following parameters can be obtained by DHCP:

Basic Configuration

IP Address

Subnet Mask (option 1)

Extended Configuration

Default route (option 3)

Static IP routing (option 33)

SNTP server (option 42)

Timezone offset (option 2)

Primary/secondary DNS server (option 6)

DNS domain name (option 15)

SIP Addresses / SIP Server & Registrar (SIP Server option 120)

Vendor unique (option 43)

The vendor specific option (code 43), or alternatively, a vendor class, is used to provide the phone with the location of an optional configuration/provisioning service. By this means, full Plug&Play is possible (see the Plug&Play) section. For further information, including an example configuration for dhcp, please refer to the
Administration Manual OpenStage Asterisk.

Plug&Play

A fully automated mass rollout of OpenStage phones and OpenScape Desk Phones can be realized by combining a DHCP server and a provisioning service which uses the WPI. On startup, the phone receives the IP address of the provisioning server from the DHCP server. After that, it contacts the provisioning service. The provisioning service may then request all settings from the phone in order to decide which parameters must be set or updated. When all these parameters have been sent to the phone, it is ready for operation.
For further information, please see the WPI article; for a deeper understanding, refer to the OpenStage Provisioning Interface Developer's Guide.

Using OpenStage Phone with Asterisk

For an overview of the features introduced with firmware version V2R1, please refer to ReadMe V2 R1 100907

Feature Table

In this table, you find information on all features which are supported by OpenStage phones connected to an Asterisk PBX.

Allows a computer to interact with the phone, e.g. in setting up and terminating calls via a PC application. OpenStage phones support 3rd party call control via SIP and uaCSTA. By the use uf uaCSTA, operations like call answering, putting a call on hold, making a call, setting microphone and speaker settings, and many more, can be performed.

Call completion is a telephony feature which takes action on a failure to complete a call. It allows for notifying the calling user when the called user is available again. CCBS (Call Completion Busy Subscriber) will take effect when the called party is busy; CCNR (Call Completion No Reply) will take effect when the called party does not respond.

Blind call transfer and call transfer with consultation are supported. In a blind transfer scenario, user A selects the blind transfer option during a conversation with user B and enters the number of user C. After that, user B is disconnected from user A and rings at user C's phone. In a consultation scenario, user A initiates a consultation call to user C during a conversation with user B. After haveing returned to the conversation with B, he selects the transfer option. User A is disconnected from user B, and B is connected to C.

-

Executive/Assistant configuration

OpenStage 60/80 ≥ V1 R5.6.0

Complex configurations with multiple executive and assistents which indicate the current status of the relevant persons can be realized using the OpenStage XML application platform.

If control codes are to be sent to the PBX during a call, DTMF (Dual Tone Multi Frequency) tones can be used.

-

Alternate Call

OpenStage 15/20/40/60/80 ≥ V1 R5.6.0

The user can alternate between the currently active call and another call that is on hold.

-

Call Hold

OpenStage 15/20/40/60/80 ≥ V1 R5.6.0

The user can put a call on hold in order to switch over to another connected call or to call another party.

-

Consultation

OpenStage 15/20/40/60/80 ≥ V1 R5.6.0

During an active call, the user can initiate a consultation call to a third party. After that, he can alternate between the two parties.

-

CLIP

OpenStage 15/20/40/60/80 ≥ V1 R5.6.0

When the call number resp. caller ID is transmitted within an incoming call, it is displayed in ringing state.

-

CLIR

OpenStage 15/20/40/60/80 ≥ V1 R5.6.0

Caller ID transmission is suppressed.

-

Local Music on Hold

OpenStage 15/20/40/60/80 ≥ V1 R5.6.0

If desired, OpenStage phones can be configured to play custom hold music to the user when put on hold. The audio or mp3 file can be uploaded via FTP; the download can be initated via local menu, WBM or WPI.

The Multiple Address Appearance feature provides the served user with multiple addresses appearing on a single telephone. The served user has the ability to originate, receive and otherwise control calls on each of these address appearances. These address appearances behave independently of each other.

Allows a computer to interact with the phone, e.g. in setting up and terminating calls via a PC application. OpenStage phones support 3rd party call control via SIP and uaCSTA. By the use uf uaCSTA, operations like call answering, putting a call on hold, making a call, setting microphone and speaker settings, and many more, can be performed.

Call completion is a telephony feature which takes action on a failure to complete a call. It allows for notifying the calling user when the called user is available again. CCBS (Call Completion Busy Subscriber) will take effect when the called party is busy; CCNR (Call Completion No Reply) will take effect when the called party does not respond.

Blind call transfer and call transfer with consultation are supported. In a blind transfer scenario, user A selects the blind transfer option during a conversation with user B and enters the number of user C. After that, user B is disconnected from user A and rings at user C's phone. In a consultation scenario, user A initiates a consultation call to user C during a conversation with user B. After haveing returned to the conversation with B, he selects the transfer option. User A is disconnected from user B, and B is connected to C.

-

Executive/Assistant configuration

OpenScape Desk Phone IP 55G ≥ V3 R2.0.0

Complex configurations with multiple executive and assistents which indicate the current status of the relevant persons can be realized using the OpenStage XML application platform.

If control codes are to be sent to the PBX during a call, DTMF (Dual Tone Multi Frequency) tones can be used.

-

Alternate Call

OpenScape Desk Phone IP 35G/55G ≥ V3 R2.0.0

The user can alternate between the currently active call and another call that is on hold.

-

Call Hold

OpenScape Desk Phone IP 35G/55G ≥ V3 R2.0.0

The user can put a call on hold in order to switch over to another connected call or to call another party.

-

Consultation

OpenScape Desk Phone IP 35G/55G ≥ V3 R2.0.0

During an active call, the user can initiate a consultation call to a third party. After that, he can alternate between the two parties.

-

CLIP

OpenScape Desk Phone IP 35G/55G ≥ V3 R2.0.0

When the call number resp. caller ID is transmitted within an incoming call, it is displayed in ringing state.

-

CLIR

OpenScape Desk Phone IP 35G/55G ≥ V3 R2.0.0

Caller ID transmission is suppressed.

-

Local Music on Hold

OpenScape Desk Phone IP 35G/55G ≥ V3 R2.0.0

If desired, OpenScape Desk Phones can be configured to play custom hold music to the user when put on hold. The audio or mp3 file can be uploaded via FTP; the download can be initated via local menu, WBM or WPI.

The Multiple Address Appearance feature provides the served user with multiple addresses appearing on a single telephone. The served user has the ability to originate, receive and otherwise control calls on each of these address appearances. These address appearances behave independently of each other.

LAN Port Mirroring

Every OpenStage phone and OpenScape Desk Phone has a built-in Ethernet switch with a LAN port and a PC port. For development and error tracing, the PC port enables network monitoring when configured as a mirror for the LAN port. For this purpose, PC port mode must be set to "mirror". If configured this way, the complete traffic of the LAN port will be passed through to the PC port, just like with a simple network hub. Now, a network tracing tool on the PC can trace all IP traffic, like SIP over UDP, or XML over HTTP, for instance.

Tracing Capabilities within the Phone

Basic Troubleshooting

For tracking network issues, the phone can execute ping and traceroute tests; these can be controlled and viewed online using the WBM.

For elementary troubleshooting, the phone provides an overview about basic issues in the user menu. The admin can ask the user to read that basic information to get a first hint about the possible causes of an issue. For a table which contains the possible error codes and their causes, please see the Error Codes section of the OpenStage SIP FAQ.

Local and Remote Tracing

The phone is able to write internal trace files, and to send the trace data to a remote syslog server. The tracing can be configured in a differentiated way by setting discrete trace levels for each service.
Please note that, order to preserver phone ressources, it is not recommended to enable all traces to the deepest level.

QoS Data Collection

OpenStage phones and OpenScape Desk Phones generate QoS reports using a HiPath specific format, QDC (QoS Data Collection).
The reports created for the last 6 sessions, i. e. conversations, can be viewed on the WBM or are reported to the QCU (QoS data Collection Unit).
SEN provides a server application to collect the data. The collected data is sent via SNMP. If an SNMP server is available, the QDC MIBS can be downloaded from our software supply server (SWS).
Meanwhile, third party solutions are available which can also deal with the OpenStage QDC data.

HUSIM Phone Tester

This tool enables the service staff to access a defined group of phones remotely.

For each phone, a PC application window shows the current status. Every OpenStage phone and OpenScape Desk Phone model is represented with its complete key layout and display content. The remote visitor can see all user interactions on the phone. Moreover, he can access the phone keys actively and in this way operate the phone by remote control. Please note that, for privacy protection, the user is always informed about the remote interaction.

To get the phone tester up and running, a special dongle key must be uploaded to the phone. The dongle key and the HUSIM software can be downloaded without additional charge from SWS/SEBA. The key can be distributed to the phone using the SEN DLS (Deployment Service) or the phone’s WPI (WorkPoint Interface).